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BUGZ-85 - handle thread safety issues with calling the decoder from the

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real-time thread.
This commit is contained in:
Roxanne Skelly 2019-05-08 16:17:18 -07:00
parent 4a4a92c009
commit dc7ec35544
6 changed files with 59 additions and 16 deletions
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1
assignment-client/src/audio/AvatarAudioStream.cpp
View file

@ -45,6 +45,7 @@ int AvatarAudioStream::parseStreamProperties(PacketType type, const QByteArray&
: AudioConstants::NETWORK_FRAME_SAMPLES_PER_CHANNEL);
// restart the codec
if (_codec) {
QMutexLocker lock(&_decoderMutex);
if (_decoder) {
_codec->releaseDecoder(_decoder);
}

2
libraries/audio/src/AudioSRC.cpp
View file

@ -1491,7 +1491,6 @@ AudioSRC::~AudioSRC() {
//
int AudioSRC::render(const int16_t* input, int16_t* output, int inputFrames) {
int outputFrames = 0;
QMutexLocker lock(&_renderMutex);
while (inputFrames) {
int ni = MIN(inputFrames, _inputBlock);
@ -1516,7 +1515,6 @@ int AudioSRC::render(const int16_t* input, int16_t* output, int inputFrames) {
//
int AudioSRC::render(const float* input, float* output, int inputFrames) {
int outputFrames = 0;
QMutexLocker lock(&_renderMutex);
while (inputFrames) {
int ni = MIN(inputFrames, _inputBlock);

2
libraries/audio/src/AudioSRC.h
View file

@ -13,7 +13,6 @@
#define hifi_AudioSRC_h
#include <stdint.h>
#include <QMutex>
static const int SRC_MAX_CHANNELS = 4;
@ -56,7 +55,6 @@ public:
int getMaxInput(int outputFrames);
private:
QMutex _renderMutex;
float* _polyphaseFilter;
int* _stepTable;

54
libraries/audio/src/InboundAudioStream.cpp
View file

@ -246,12 +246,20 @@ int InboundAudioStream::lostAudioData(int numPackets) {
QByteArray decodedBuffer;
while (numPackets--) {
if (!_decoderMutex.tryLock()) {
// an incoming packet is being processed,
// and will likely be on the ring buffer shortly,
// so don't bother generating more data
qCInfo(audiostream, "Packet currently being unpacked or lost frame already being generated. Not generating lost frame.");
return 0;
}
if (_decoder) {
_decoder->lostFrame(decodedBuffer);
} else {
decodedBuffer.resize(AudioConstants::NETWORK_FRAME_BYTES_PER_CHANNEL * _numChannels);
memset(decodedBuffer.data(), 0, decodedBuffer.size());
}
_decoderMutex.unlock();
_ringBuffer.writeData(decodedBuffer.data(), decodedBuffer.size());
}
return 0;
@ -259,6 +267,12 @@ int InboundAudioStream::lostAudioData(int numPackets) {
int InboundAudioStream::parseAudioData(PacketType type, const QByteArray& packetAfterStreamProperties) {
QByteArray decodedBuffer;
// may block on the real-time thread, which is acceptible as
// parseAudioData is only called by the packet processing
// thread which, while high performance, is not as sensitive to
// delays as the real-time thread.
QMutexLocker lock(&_decoderMutex);
if (_decoder) {
_decoder->decode(packetAfterStreamProperties, decodedBuffer);
} else {
@ -277,16 +291,23 @@ int InboundAudioStream::writeDroppableSilentFrames(int silentFrames) {
// case we will call the decoder's lostFrame() method, which indicates
// that it should interpolate from its last known state down toward
// silence.
if (_decoder) {
// FIXME - We could potentially use the output from the codec, in which
// case we might get a cleaner fade toward silence. NOTE: The below logic
// attempts to catch up in the event that the jitter buffers have grown.
// The better long term fix is to use the output from the decode, detect
// when it actually reaches silence, and then delete the silent portions
// of the jitter buffers. Or petentially do a cross fade from the decode
// output to silence.
QByteArray decodedBuffer;
_decoder->lostFrame(decodedBuffer);
{
// may block on the real-time thread, which is acceptible as
// writeDroppableSilentFrames is only called by the packet processing
// thread which, while high performance, is not as sensitive to
// delays as the real-time thread.
QMutexLocker lock(&_decoderMutex);
if (_decoder) {
// FIXME - We could potentially use the output from the codec, in which
// case we might get a cleaner fade toward silence. NOTE: The below logic
// attempts to catch up in the event that the jitter buffers have grown.
// The better long term fix is to use the output from the decode, detect
// when it actually reaches silence, and then delete the silent portions
// of the jitter buffers. Or petentially do a cross fade from the decode
// output to silence.
QByteArray decodedBuffer;
_decoder->lostFrame(decodedBuffer);
}
}
// calculate how many silent frames we should drop.
@ -344,7 +365,16 @@ int InboundAudioStream::popSamples(int maxSamples, bool allOrNothing) {
//Kick PLC to generate a filler frame, reducing 'click'
lostAudioData(allOrNothing ? (maxSamples - samplesAvailable) / _ringBuffer.getNumFrameSamples() : 1);
samplesPopped = _ringBuffer.samplesAvailable();
popSamplesNoCheck(samplesPopped);
if (samplesPopped) {
popSamplesNoCheck(samplesPopped);
} else {
// No samples available means a packet is currently being
// processed, so we don't generate lost audio data, and instead
// just wait for the packet to come in. This prevents locking
// the real-time audio thread at the cost of a potential (but rare)
// 'click'
_lastPopSucceeded = false;
}
}
}
return samplesPopped;
@ -531,6 +561,7 @@ void InboundAudioStream::setupCodec(CodecPluginPointer codec, const QString& cod
_codec = codec;
_selectedCodecName = codecName;
if (_codec) {
QMutexLocker lock(&_decoderMutex);
_decoder = codec->createDecoder(AudioConstants::SAMPLE_RATE, numChannels);
}
}
@ -538,6 +569,7 @@ void InboundAudioStream::setupCodec(CodecPluginPointer codec, const QString& cod
void InboundAudioStream::cleanupCodec() {
// release any old codec encoder/decoder first...
if (_codec) {
QMutexLocker lock(&_decoderMutex);
if (_decoder) {
_codec->releaseDecoder(_decoder);
_decoder = nullptr;

1
libraries/audio/src/InboundAudioStream.h
View file

@ -187,6 +187,7 @@ protected:
CodecPluginPointer _codec;
QString _selectedCodecName;
QMutex _decoderMutex;
Decoder* _decoder { nullptr };
int _mismatchedAudioCodecCount { 0 };
};

15
libraries/audio/src/MixedProcessedAudioStream.cpp
View file

@ -36,13 +36,20 @@ int MixedProcessedAudioStream::lostAudioData(int numPackets) {
QByteArray outputBuffer;
while (numPackets--) {
if (!_decoderMutex.tryLock()) {
// an incoming packet is being processed,
// and will likely be on the ring buffer shortly,
// so don't bother generating more data
qCInfo(audiostream, "Packet currently being unpacked or lost frame already being generated. Not generating lost frame.");
return 0;
}
if (_decoder) {
_decoder->lostFrame(decodedBuffer);
} else {
decodedBuffer.resize(AudioConstants::NETWORK_FRAME_BYTES_STEREO);
memset(decodedBuffer.data(), 0, decodedBuffer.size());
}
_decoderMutex.unlock();
emit addedStereoSamples(decodedBuffer);
emit processSamples(decodedBuffer, outputBuffer);
@ -55,6 +62,12 @@ int MixedProcessedAudioStream::lostAudioData(int numPackets) {
int MixedProcessedAudioStream::parseAudioData(PacketType type, const QByteArray& packetAfterStreamProperties) {
QByteArray decodedBuffer;
// may block on the real-time thread, which is acceptible as
// parseAudioData is only called by the packet processing
// thread which, while high performance, is not as sensitive to
// delays as the real-time thread.
QMutexLocker lock(&_decoderMutex);
if (_decoder) {
_decoder->decode(packetAfterStreamProperties, decodedBuffer);
} else {